diff --git a/dsd/dsd_audio.c b/dsd/dsd_audio.c
index 8c15948b3..75f109ad7 100644
--- a/dsd/dsd_audio.c
+++ b/dsd/dsd_audio.c
@@ -17,424 +17,433 @@
 
 #include "dsd.h"
 
-void
-processAudio (dsd_opts * opts, dsd_state * state)
+void processAudio(dsd_opts * opts, dsd_state * state)
 {
 
-  int i, n;
-  float aout_abs, max, gainfactor, gaindelta, maxbuf;
+    int i, n;
+    float aout_abs, max, gainfactor, gaindelta, maxbuf;
 
-  if (opts->audio_gain == (float) 0)
+    if (opts->audio_gain == (float) 0)
     {
-      // detect max level
-      max = 0;
-      state->audio_out_temp_buf_p = state->audio_out_temp_buf;
-      for (n = 0; n < 160; n++)
+        // detect max level
+        max = 0;
+        state->audio_out_temp_buf_p = state->audio_out_temp_buf;
+        for (n = 0; n < 160; n++)
         {
-          aout_abs = fabsf (*state->audio_out_temp_buf_p);
-          if (aout_abs > max)
+            aout_abs = fabsf(*state->audio_out_temp_buf_p);
+            if (aout_abs > max)
             {
-              max = aout_abs;
+                max = aout_abs;
             }
-          state->audio_out_temp_buf_p++;
+            state->audio_out_temp_buf_p++;
         }
-      *state->aout_max_buf_p = max;
-      state->aout_max_buf_p++;
-      state->aout_max_buf_idx++;
-      if (state->aout_max_buf_idx > 24)
+        *state->aout_max_buf_p = max;
+        state->aout_max_buf_p++;
+        state->aout_max_buf_idx++;
+        if (state->aout_max_buf_idx > 24)
         {
-          state->aout_max_buf_idx = 0;
-          state->aout_max_buf_p = state->aout_max_buf;
+            state->aout_max_buf_idx = 0;
+            state->aout_max_buf_p = state->aout_max_buf;
         }
 
-      // lookup max history
-      for (i = 0; i < 25; i++)
+        // lookup max history
+        for (i = 0; i < 25; i++)
         {
-          maxbuf = state->aout_max_buf[i];
-          if (maxbuf > max)
+            maxbuf = state->aout_max_buf[i];
+            if (maxbuf > max)
             {
-              max = maxbuf;
+                max = maxbuf;
             }
         }
 
-      // determine optimal gain level
-      if (max > (float) 0)
+        // determine optimal gain level
+        if (max > (float) 0)
         {
-          gainfactor = ((float) 30000 / max);
+            gainfactor = ((float) 30000 / max);
         }
-      else
+        else
         {
-          gainfactor = (float) 50;
+            gainfactor = (float) 50;
         }
-      if (gainfactor < state->aout_gain)
+        if (gainfactor < state->aout_gain)
         {
-          state->aout_gain = gainfactor;
-          gaindelta = (float) 0;
+            state->aout_gain = gainfactor;
+            gaindelta = (float) 0;
         }
-      else
+        else
         {
-          if (gainfactor > (float) 50)
+            if (gainfactor > (float) 50)
             {
-              gainfactor = (float) 50;
+                gainfactor = (float) 50;
             }
-          gaindelta = gainfactor - state->aout_gain;
-          if (gaindelta > ((float) 0.05 * state->aout_gain))
+            gaindelta = gainfactor - state->aout_gain;
+            if (gaindelta > ((float) 0.05 * state->aout_gain))
             {
-              gaindelta = ((float) 0.05 * state->aout_gain);
+                gaindelta = ((float) 0.05 * state->aout_gain);
             }
         }
-      gaindelta /= (float) 160;
+        gaindelta /= (float) 160;
     }
-  else
+    else
     {
-      gaindelta = (float) 0;
+        gaindelta = (float) 0;
     }
 
-  if(opts->audio_gain >= 0){
-    // adjust output gain
+    if (opts->audio_gain >= 0)
+    {
+        // adjust output gain
+        state->audio_out_temp_buf_p = state->audio_out_temp_buf;
+        for (n = 0; n < 160; n++)
+        {
+            *state->audio_out_temp_buf_p = (state->aout_gain
+                    + ((float) n * gaindelta)) * (*state->audio_out_temp_buf_p);
+            state->audio_out_temp_buf_p++;
+        }
+        state->aout_gain += ((float) 160 * gaindelta);
+    }
+
+    // copy audio datat to output buffer and upsample if necessary
     state->audio_out_temp_buf_p = state->audio_out_temp_buf;
-    for (n = 0; n < 160; n++)
-      {
-        *state->audio_out_temp_buf_p = (state->aout_gain + ((float) n * gaindelta)) * (*state->audio_out_temp_buf_p);
-        state->audio_out_temp_buf_p++;
-      }
-    state->aout_gain += ((float) 160 * gaindelta);
-  }
-
-  // copy audio datat to output buffer and upsample if necessary
-  state->audio_out_temp_buf_p = state->audio_out_temp_buf;
-  if (opts->split == 0)
+    if ((opts->split == 0) || (opts->upsample))
     {
-      for (n = 0; n < 160; n++)
+        for (n = 0; n < 160; n++)
         {
-          upsample (state, *state->audio_out_temp_buf_p);
-          state->audio_out_temp_buf_p++;
-          state->audio_out_float_buf_p += 6;
-          state->audio_out_idx += 6;
-          state->audio_out_idx2 += 6;
+            upsample(state, *state->audio_out_temp_buf_p);
+            state->audio_out_temp_buf_p++;
+            state->audio_out_float_buf_p += 6;
+            state->audio_out_idx += 6;
+            state->audio_out_idx2 += 6;
         }
-      state->audio_out_float_buf_p -= (960 + opts->playoffset);
-      // copy to output (short) buffer
-      for (n = 0; n < 960; n++)
+        state->audio_out_float_buf_p -= (960 + opts->playoffset);
+        // copy to output (short) buffer
+        for (n = 0; n < 960; n++)
         {
-          if (*state->audio_out_float_buf_p > (float) 32760)
+            if (*state->audio_out_float_buf_p > (float) 32760)
             {
-              *state->audio_out_float_buf_p = (float) 32760;
+                *state->audio_out_float_buf_p = (float) 32760;
             }
-          else if (*state->audio_out_float_buf_p < (float) -32760)
+            else if (*state->audio_out_float_buf_p < (float) -32760)
             {
-              *state->audio_out_float_buf_p = (float) -32760;
+                *state->audio_out_float_buf_p = (float) -32760;
             }
-          *state->audio_out_buf_p = (short) *state->audio_out_float_buf_p;
-          state->audio_out_buf_p++;
-          state->audio_out_float_buf_p++;
+            *state->audio_out_buf_p = (short) *state->audio_out_float_buf_p;
+            state->audio_out_buf_p++;
+            state->audio_out_float_buf_p++;
         }
-      state->audio_out_float_buf_p += opts->playoffset;
+        state->audio_out_float_buf_p += opts->playoffset;
     }
-  else
+    else
     {
-      for (n = 0; n < 160; n++)
+        for (n = 0; n < 160; n++)
         {
-          if (*state->audio_out_temp_buf_p > (float) 32760)
+            if (*state->audio_out_temp_buf_p > (float) 32760)
             {
-              *state->audio_out_temp_buf_p = (float) 32760;
+                *state->audio_out_temp_buf_p = (float) 32760;
             }
-          else if (*state->audio_out_temp_buf_p < (float) -32760)
+            else if (*state->audio_out_temp_buf_p < (float) -32760)
             {
-              *state->audio_out_temp_buf_p = (float) -32760;
+                *state->audio_out_temp_buf_p = (float) -32760;
             }
-          *state->audio_out_buf_p = (short) *state->audio_out_temp_buf_p;
-          state->audio_out_buf_p++;
-          state->audio_out_temp_buf_p++;
-          state->audio_out_idx++;
-          state->audio_out_idx2++;
+            *state->audio_out_buf_p = (short) *state->audio_out_temp_buf_p;
+            state->audio_out_buf_p++;
+            state->audio_out_temp_buf_p++;
+            state->audio_out_idx++;
+            state->audio_out_idx2++;
         }
     }
 }
 
-void
-writeSynthesizedVoice (dsd_opts * opts, dsd_state * state)
+void writeSynthesizedVoice(dsd_opts * opts, dsd_state * state)
 {
 #ifdef USE_LIBSNDFILE
-  int n;
-  short aout_buf[160];
-  short *aout_buf_p;
+    int n;
+    short aout_buf[160];
+    short *aout_buf_p;
 
 //  for(n=0; n<160; n++)
 //    fprintf(stderr, "%d ", ((short*)(state->audio_out_temp_buf))[n]);
 //  fprintf(stderr, "\n");
 
-  aout_buf_p = aout_buf;
-  state->audio_out_temp_buf_p = state->audio_out_temp_buf;
+    aout_buf_p = aout_buf;
+    state->audio_out_temp_buf_p = state->audio_out_temp_buf;
 
-  for (n = 0; n < 160; n++)
-  {
-    if (*state->audio_out_temp_buf_p > (float) 32767)
-      {
-        *state->audio_out_temp_buf_p = (float) 32767;
-      }
-    else if (*state->audio_out_temp_buf_p < (float) -32768)
-      {
-        *state->audio_out_temp_buf_p = (float) -32768;
-      }
-      *aout_buf_p = (short) *state->audio_out_temp_buf_p;
-      aout_buf_p++;
-      state->audio_out_temp_buf_p++;
-  }
-
-  sf_write_short(opts->wav_out_f, aout_buf, 160);
-  /*
-
-  int n;
-  short aout_buf[160];
-  short *aout_buf_p;
-  ssize_t result;
-
-  aout_buf_p = aout_buf;
-  state->audio_out_temp_buf_p = state->audio_out_temp_buf;
-  for (n = 0; n < 160; n++)
+    for (n = 0; n < 160; n++)
     {
-      if (*state->audio_out_temp_buf_p > (float) 32760)
+        if (*state->audio_out_temp_buf_p > (float) 32767)
         {
-          *state->audio_out_temp_buf_p = (float) 32760;
+            *state->audio_out_temp_buf_p = (float) 32767;
         }
-      else if (*state->audio_out_temp_buf_p < (float) -32760)
+        else if (*state->audio_out_temp_buf_p < (float) -32768)
         {
-          *state->audio_out_temp_buf_p = (float) -32760;
+            *state->audio_out_temp_buf_p = (float) -32768;
         }
-      *aout_buf_p = (short) *state->audio_out_temp_buf_p;
-      aout_buf_p++;
-      state->audio_out_temp_buf_p++;
+        *aout_buf_p = (short) *state->audio_out_temp_buf_p;
+        aout_buf_p++;
+        state->audio_out_temp_buf_p++;
     }
 
-  result = write (opts->wav_out_fd, aout_buf, 320);
-  fflush (opts->wav_out_f);
-  state->wav_out_bytes += 320;
-  */
+    sf_write_short(opts->wav_out_f, aout_buf, 160);
+    /*
+
+     int n;
+     short aout_buf[160];
+     short *aout_buf_p;
+     ssize_t result;
+
+     aout_buf_p = aout_buf;
+     state->audio_out_temp_buf_p = state->audio_out_temp_buf;
+     for (n = 0; n < 160; n++)
+     {
+     if (*state->audio_out_temp_buf_p > (float) 32760)
+     {
+     *state->audio_out_temp_buf_p = (float) 32760;
+     }
+     else if (*state->audio_out_temp_buf_p < (float) -32760)
+     {
+     *state->audio_out_temp_buf_p = (float) -32760;
+     }
+     *aout_buf_p = (short) *state->audio_out_temp_buf_p;
+     aout_buf_p++;
+     state->audio_out_temp_buf_p++;
+     }
+
+     result = write (opts->wav_out_fd, aout_buf, 320);
+     fflush (opts->wav_out_f);
+     state->wav_out_bytes += 320;
+     */
 #endif
 }
 
-void
-playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
+void playSynthesizedVoice(dsd_opts * opts, dsd_state * state)
 {
 
-  ssize_t result;
+    ssize_t result;
 
-  if (state->audio_out_idx > opts->delay)
+    if (state->audio_out_idx > opts->delay)
     {
-      // output synthesized speech to sound card
-      if (opts->audio_out_fd == -1)
+        // output synthesized speech to sound card
+        if (opts->audio_out_fd == -1)
         {
-          memcpy(state->output_buffer + state->output_offset, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2));
-          state->output_offset += state->audio_out_idx;
+            memcpy(state->output_buffer + state->output_offset,
+                    (state->audio_out_buf_p - state->audio_out_idx),
+                    (state->audio_out_idx * 2));
+            state->output_offset += state->audio_out_idx;
         }
-      else
+        else
         {
-          result = write (opts->audio_out_fd, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2));
+            result = write(opts->audio_out_fd,
+                    (state->audio_out_buf_p - state->audio_out_idx),
+                    (state->audio_out_idx * 2));
         }
-      state->audio_out_idx = 0;
+        state->audio_out_idx = 0;
     }
 
-  if (state->audio_out_idx2 >= 800000)
+    if (state->audio_out_idx2 >= 800000)
     {
-      state->audio_out_float_buf_p = state->audio_out_float_buf + 100;
-      state->audio_out_buf_p = state->audio_out_buf + 100;
-      memset (state->audio_out_float_buf, 0, 100 * sizeof (float));
-      memset (state->audio_out_buf, 0, 100 * sizeof (short));
-      state->audio_out_idx2 = 0;
+        state->audio_out_float_buf_p = state->audio_out_float_buf + 100;
+        state->audio_out_buf_p = state->audio_out_buf + 100;
+        memset(state->audio_out_float_buf, 0, 100 * sizeof(float));
+        memset(state->audio_out_buf, 0, 100 * sizeof(short));
+        state->audio_out_idx2 = 0;
     }
 }
 
-void
-openAudioOutDevice (dsd_opts * opts, int speed)
+void openAudioOutDevice(dsd_opts * opts, int speed)
 {
-  // get info of device/file
-  struct stat stat_buf;
-  if(stat(opts->audio_out_dev, &stat_buf) != 0) {
-    fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
-    exit(1);
-  }
-
-  if( !(S_ISCHR(stat_buf.st_mode) || S_ISBLK(stat_buf.st_mode))) { // this is not a device
-    fprintf(stderr, "Error, %s is not a device. use -w filename for wav output.\n", opts->audio_out_dev);
-    exit(1);
-  }
-#ifdef SOLARIS
-  sample_info_t aset, aget;
-
-  opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
-  if (opts->audio_out_fd == -1)
+    // get info of device/file
+    struct stat stat_buf;
+    if (stat(opts->audio_out_dev, &stat_buf) != 0)
     {
-      fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
-      exit (1);
-    }
-
-  // get current
-  ioctl (opts->audio_out_fd, AUDIO_GETINFO, &aset);
-
-  aset.record.sample_rate = speed;
-  aset.play.sample_rate = speed;
-  aset.record.channels = 1;
-  aset.play.channels = 1;
-  aset.record.precision = 16;
-  aset.play.precision = 16;
-  aset.record.encoding = AUDIO_ENCODING_LINEAR;
-  aset.play.encoding = AUDIO_ENCODING_LINEAR;
-
-  if (ioctl (opts->audio_out_fd, AUDIO_SETINFO, &aset) == -1)
-    {
-      fprintf(stderr, "Error setting sample device parameters\n");
-      exit (1);
-    }
-#endif
-
-#if defined(BSD) && !defined(__APPLE__) && defined(USE_LIBSNDFILE)
-
-  int fmt;
-
-  opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
-  if (opts->audio_out_fd == -1)
-    {
-      fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
-      opts->audio_out = 0;
-      exit(1);
-    }
-
-  fmt = 0;
-  if (ioctl (opts->audio_out_fd, SNDCTL_DSP_RESET) < 0)
-    {
-      fprintf(stderr, "ioctl reset error \n");
-    }
-  fmt = speed;
-  if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
-    {
-      fprintf(stderr, "ioctl speed error \n");
-    }
-  fmt = 0;
-  if (ioctl (opts->audio_out_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
-    {
-      fprintf(stderr, "ioctl stereo error \n");
-    }
-  fmt = AFMT_S16_LE;
-  if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
-    {
-      fprintf(stderr, "ioctl setfmt error \n");
-    }
-
-#endif
-  fprintf(stderr, "Audio Out Device: %s\n", opts->audio_out_dev);
-}
-
-void
-openAudioInDevice (dsd_opts * opts)
-{
-#ifdef USE_LIBSNDFILE
-  // get info of device/file
-  struct stat stat_buf;
-  if (stat(opts->audio_in_dev, &stat_buf) != 0) {
-    fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
-    exit(1);
-  }
-  if(S_ISREG(stat_buf.st_mode)) { // is this a regular file? then process with libsndfile.
-    opts->audio_in_type = 1;
-    opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
-    opts->audio_in_file_info->channels = 1;
-    opts->audio_in_file = sf_open(opts->audio_in_dev, SFM_READ, opts->audio_in_file_info);
-    if(opts->audio_in_file == NULL) {
-        fprintf(stderr, "Error, couldn't open file %s\n", opts->audio_in_dev);
+        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
+        exit(1);
+    }
+
+    if (!(S_ISCHR(stat_buf.st_mode) || S_ISBLK(stat_buf.st_mode)))
+    { // this is not a device
+        fprintf(stderr,
+                "Error, %s is not a device. use -w filename for wav output.\n",
+                opts->audio_out_dev);
         exit(1);
     }
-  }
-  else { // this is a device, use old handling
-  opts->audio_in_type = 0;
 #ifdef SOLARIS
     sample_info_t aset, aget;
-    int rgain;
 
-    rgain = 64;
-
-    if (opts->split == 1)
-      {
-        opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
-      }
-    else
-      {
-        opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
-      }
-    if (opts->audio_in_fd == -1)
-      {
-        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
-        exit(1);
-      }
+    opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
+    if (opts->audio_out_fd == -1)
+    {
+        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
+        exit (1);
+    }
 
     // get current
-    ioctl (opts->audio_in_fd, AUDIO_GETINFO, &aset);
+    ioctl (opts->audio_out_fd, AUDIO_GETINFO, &aset);
 
-    aset.record.sample_rate = 48000;
-    aset.play.sample_rate = 48000;
+    aset.record.sample_rate = speed;
+    aset.play.sample_rate = speed;
     aset.record.channels = 1;
     aset.play.channels = 1;
     aset.record.precision = 16;
     aset.play.precision = 16;
     aset.record.encoding = AUDIO_ENCODING_LINEAR;
     aset.play.encoding = AUDIO_ENCODING_LINEAR;
-    aset.record.port = AUDIO_LINE_IN;
-    aset.record.gain = rgain;
 
-    if (ioctl (opts->audio_in_fd, AUDIO_SETINFO, &aset) == -1)
-      {
+    if (ioctl (opts->audio_out_fd, AUDIO_SETINFO, &aset) == -1)
+    {
         fprintf(stderr, "Error setting sample device parameters\n");
         exit (1);
-      }
+    }
+#endif
+
+#if defined(BSD) && !defined(__APPLE__) && defined(USE_LIBSNDFILE)
+
+    int fmt;
+
+    opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
+    if (opts->audio_out_fd == -1)
+    {
+        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_out_dev);
+        opts->audio_out = 0;
+        exit(1);
+    }
+
+    fmt = 0;
+    if (ioctl (opts->audio_out_fd, SNDCTL_DSP_RESET) < 0)
+    {
+        fprintf(stderr, "ioctl reset error \n");
+    }
+    fmt = speed;
+    if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
+    {
+        fprintf(stderr, "ioctl speed error \n");
+    }
+    fmt = 0;
+    if (ioctl (opts->audio_out_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
+    {
+        fprintf(stderr, "ioctl stereo error \n");
+    }
+    fmt = AFMT_S16_LE;
+    if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
+    {
+        fprintf(stderr, "ioctl setfmt error \n");
+    }
+
+#endif
+    fprintf(stderr, "Audio Out Device: %s\n", opts->audio_out_dev);
+}
+
+void openAudioInDevice(dsd_opts * opts)
+{
+#ifdef USE_LIBSNDFILE
+    // get info of device/file
+    struct stat stat_buf;
+    if (stat(opts->audio_in_dev, &stat_buf) != 0)
+    {
+        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
+        exit(1);
+    }
+    if(S_ISREG(stat_buf.st_mode))
+    { // is this a regular file? then process with libsndfile.
+        opts->audio_in_type = 1;
+        opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
+        opts->audio_in_file_info->channels = 1;
+        opts->audio_in_file = sf_open(opts->audio_in_dev, SFM_READ, opts->audio_in_file_info);
+        if(opts->audio_in_file == NULL)
+        {
+            fprintf(stderr, "Error, couldn't open file %s\n", opts->audio_in_dev);
+            exit(1);
+        }
+    }
+    else
+    { // this is a device, use old handling
+        opts->audio_in_type = 0;
+#ifdef SOLARIS
+        sample_info_t aset, aget;
+        int rgain;
+
+        rgain = 64;
+
+        if (opts->split == 1)
+        {
+            opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
+        }
+        else
+        {
+            opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
+        }
+        if (opts->audio_in_fd == -1)
+        {
+            fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
+            exit(1);
+        }
+
+        // get current
+        ioctl (opts->audio_in_fd, AUDIO_GETINFO, &aset);
+
+        aset.record.sample_rate = 48000;
+        aset.play.sample_rate = 48000;
+        aset.record.channels = 1;
+        aset.play.channels = 1;
+        aset.record.precision = 16;
+        aset.play.precision = 16;
+        aset.record.encoding = AUDIO_ENCODING_LINEAR;
+        aset.play.encoding = AUDIO_ENCODING_LINEAR;
+        aset.record.port = AUDIO_LINE_IN;
+        aset.record.gain = rgain;
+
+        if (ioctl (opts->audio_in_fd, AUDIO_SETINFO, &aset) == -1)
+        {
+            fprintf(stderr, "Error setting sample device parameters\n");
+            exit (1);
+        }
 #endif
 
 #if defined(BSD) && !defined(__APPLE__)
-    int fmt;
+        int fmt;
 
-    if (opts->split == 1)
-      {
-        opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
-      }
-    else
-      {
-        opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
-      }
+        if (opts->split == 1)
+        {
+            opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
+        }
+        else
+        {
+            opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
+        }
 
-    if (opts->audio_in_fd == -1)
-      {
-        fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
-        opts->audio_out = 0;
-      }
+        if (opts->audio_in_fd == -1)
+        {
+            fprintf(stderr, "Error, couldn't open %s\n", opts->audio_in_dev);
+            opts->audio_out = 0;
+        }
 
-    fmt = 0;
-    if (ioctl (opts->audio_in_fd, SNDCTL_DSP_RESET) < 0)
-      {
-        fprintf(stderr, "ioctl reset error \n");
-      }
-    fmt = 48000;
-    if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
-      {
-        fprintf(stderr, "ioctl speed error \n");
-      }
-    fmt = 0;
-    if (ioctl (opts->audio_in_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
-      {
-        fprintf(stderr, "ioctl stereo error \n");
-      }
-    fmt = AFMT_S16_LE;
-    if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
-      {
-        fprintf(stderr, "ioctl setfmt error \n");
-      }
+        fmt = 0;
+        if (ioctl (opts->audio_in_fd, SNDCTL_DSP_RESET) < 0)
+        {
+            fprintf(stderr, "ioctl reset error \n");
+        }
+        fmt = 48000;
+        if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
+        {
+            fprintf(stderr, "ioctl speed error \n");
+        }
+        fmt = 0;
+        if (ioctl (opts->audio_in_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
+        {
+            fprintf(stderr, "ioctl stereo error \n");
+        }
+        fmt = AFMT_S16_LE;
+        if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
+        {
+            fprintf(stderr, "ioctl setfmt error \n");
+        }
 #endif
-  }
-  if (opts->split == 1)
-    {
-      fprintf(stderr, "Audio In Device: %s\n", opts->audio_in_dev);
     }
-  else
+    if (opts->split == 1)
     {
-      fprintf(stderr, "Audio In/Out Device: %s\n", opts->audio_in_dev);
+        fprintf(stderr, "Audio In Device: %s\n", opts->audio_in_dev);
+    }
+    else
+    {
+        fprintf(stderr, "Audio In/Out Device: %s\n", opts->audio_in_dev);
     }
 #endif
 }
diff --git a/dsd/dsd_opts.c b/dsd/dsd_opts.c
index 5cecf3926..1aae0c905 100644
--- a/dsd/dsd_opts.c
+++ b/dsd/dsd_opts.c
@@ -78,5 +78,6 @@ void initOpts(dsd_opts * opts)
     opts->delay = 0;
     opts->use_cosine_filter = 1;
     opts->unmute_encrypted_p25 = 0;
+    opts->upsample = 0; // by default stay with 8k audio
 }
 
diff --git a/dsd/dsd_state.c b/dsd/dsd_state.c
index 236c29fd8..48dc08a61 100644
--- a/dsd/dsd_state.c
+++ b/dsd/dsd_state.c
@@ -132,5 +132,6 @@ void initState(dsd_state * state)
     }
 
     state->dsd_running = 0;
+    state->output_phasor = 0;
 }
 
diff --git a/dsd/dsd_state.h b/dsd/dsd_state.h
index 75b94a909..e64f71274 100644
--- a/dsd/dsd_state.h
+++ b/dsd/dsd_state.h
@@ -111,9 +111,11 @@ typedef struct
   short *output_buffer;          //!< Output of decoder single S16LE
   int output_offset;             //!< producer pointer
   short *output_samples;         //!< L+R channels S16LE ready for writing to audio FIFO
+  int output_buffers_size;       //!< Size of buffers in number of samples
   int output_num_samples;        //!< Number of L+R samples available in the above buffer
-  int output_length;             //!< L+R buffer size (fixed)
+  int output_length;             //!< Number of wished samples
   int output_finished;           //!< 0: not ready, 1: ready
+  float output_phasor;
 
   pthread_mutex_t input_mutex;
   pthread_cond_t input_ready;
diff --git a/dsd/dsd_symbol.c b/dsd/dsd_symbol.c
index 2bdbda3e6..da68809bb 100644
--- a/dsd/dsd_symbol.c
+++ b/dsd/dsd_symbol.c
@@ -102,22 +102,36 @@ int getSymbol(dsd_opts * opts, dsd_state * state, int have_sync)
                 {
                     int i;
                     state->input_length = 0; // states all samples have been consumed
+                    state->input_offset = 0;
+
+                    // debug ...
+                    for (i = 0; i < state->output_length; i++)
+                    {
+                        float s = sin(state->output_phasor);
+                        state->output_phasor = fmod(state->output_phasor + (M_PI / 48.0), 2.0 * M_PI);
+                        state->output_samples[2*state->output_offset + 2*i] = s * 16368.0f;
+                        state->output_samples[2*state->output_offset + 2*i+1] = s * 16368.0f;
+                    }
+                    state->output_offset += state->output_length;
                     state->output_num_samples = state->output_offset;
+                    // ... debug
 
-                    //ffprintf(stderr, stderr, "dsd::getSymbol: input processing has finished\n");
-
-                    if (state->output_num_samples > state->output_length)
-                    {
-                        fprintf(stderr, "WARNING: audio buffer over-run! Truncating output\n");
-                        state->output_num_samples = state->output_length;
-                    }
-
-                    for (i = 0; i < state->output_num_samples; i++)
-                    {
-                        state->output_samples[2*i] = state->output_buffer[i];    // L channel
-                        state->output_samples[2*i+1] = state->output_buffer[i];  // R channel
-                    }
-
+//                    state->output_num_samples = state->output_offset;
+//
+//                    //ffprintf(stderr, stderr, "dsd::getSymbol: input processing has finished\n");
+//
+//                    if (state->output_num_samples > state->output_length)
+//                    {
+//                        fprintf(stderr, "WARNING: audio buffer over-run! Truncating output\n");
+//                        state->output_num_samples = state->output_length;
+//                    }
+//
+//                    for (i = 0; i < state->output_num_samples; i++)
+//                    {
+//                        state->output_samples[2*i] = state->output_buffer[i];    // L channel
+//                        state->output_samples[2*i+1] = state->output_buffer[i];  // R channel
+//                    }
+//
                     state->output_finished = 1;
                 }
                 else
diff --git a/dsd/dsd_upsample.c b/dsd/dsd_upsample.c
index 3bbc584df..8caa157db 100644
--- a/dsd/dsd_upsample.c
+++ b/dsd/dsd_upsample.c
@@ -17,52 +17,100 @@
 
 #include "dsd.h"
 
-void
-upsample (dsd_state * state, float invalue)
+void upsample(dsd_state * state, float invalue)
 {
 
-  int i, j, sum;
-  float *outbuf1, c, d;
+    int i, j, sum;
+    float *outbuf1, c, d;
 
-  outbuf1 = state->audio_out_float_buf_p;
-  outbuf1--;
-  c = *outbuf1;
-  d = invalue;
-  // basic triangle interpolation
-  outbuf1++;
-  *outbuf1 = ((invalue * (float) 0.166) + (c * (float) 0.834));
-  outbuf1++;
-  *outbuf1 = ((invalue * (float) 0.332) + (c * (float) 0.668));
-  outbuf1++;
-  *outbuf1 = ((invalue * (float) 0.5) + (c * (float) 0.5));
-  outbuf1++;
-  *outbuf1 = ((invalue * (float) 0.668) + (c * (float) 0.332));
-  outbuf1++;
-  *outbuf1 = ((invalue * (float) 0.834) + (c * (float) 0.166));
-  outbuf1++;
-  *outbuf1 = d;
-  outbuf1++;
+    outbuf1 = state->audio_out_float_buf_p;
+    outbuf1--;
+    c = *outbuf1;
+    d = invalue;
+    // basic triangle interpolation
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.166) + (c * (float) 0.834));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.332) + (c * (float) 0.668));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.5) + (c * (float) 0.5));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.668) + (c * (float) 0.332));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.834) + (c * (float) 0.166));
+    outbuf1++;
+    *outbuf1 = d;
+    outbuf1++;
 
-  if (state->audio_out_idx2 > 24)
+    if (state->audio_out_idx2 > 24)
     {
-      // smoothing
-      outbuf1 -= 16;
-      for (j = 0; j < 4; j++)
+        // smoothing
+        outbuf1 -= 16;
+        for (j = 0; j < 4; j++)
         {
-          for (i = 0; i < 6; i++)
+            for (i = 0; i < 6; i++)
             {
-              sum = 0;
-              outbuf1 -= 2;
-              sum += *outbuf1;
-              outbuf1 += 2;
-              sum += *outbuf1;
-              outbuf1 += 2;
-              sum += *outbuf1;
-              outbuf1 -= 2;
-              *outbuf1 = (sum / (float) 3);
-              outbuf1++;
+                sum = 0;
+                outbuf1 -= 2;
+                sum += *outbuf1;
+                outbuf1 += 2;
+                sum += *outbuf1;
+                outbuf1 += 2;
+                sum += *outbuf1;
+                outbuf1 -= 2;
+                *outbuf1 = (sum / (float) 3);
+                outbuf1++;
             }
-          outbuf1 -= 8;
+            outbuf1 -= 8;
+        }
+    }
+}
+
+void upsample16(dsd_state * state, short invalue)
+{
+
+    int i, j, sum;
+    float *outbuf1, c, d;
+
+    outbuf1 = state->audio_out_float_buf_p;
+    outbuf1--;
+    c = *outbuf1;
+    d = invalue;
+    // basic triangle interpolation
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.166) + (c * (float) 0.834));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.332) + (c * (float) 0.668));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.5) + (c * (float) 0.5));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.668) + (c * (float) 0.332));
+    outbuf1++;
+    *outbuf1 = ((invalue * (float) 0.834) + (c * (float) 0.166));
+    outbuf1++;
+    *outbuf1 = d;
+    outbuf1++;
+
+    if (state->audio_out_idx2 > 24)
+    {
+        // smoothing
+        outbuf1 -= 16;
+        for (j = 0; j < 4; j++)
+        {
+            for (i = 0; i < 6; i++)
+            {
+                sum = 0;
+                outbuf1 -= 2;
+                sum += *outbuf1;
+                outbuf1 += 2;
+                sum += *outbuf1;
+                outbuf1 += 2;
+                sum += *outbuf1;
+                outbuf1 -= 2;
+                *outbuf1 = (sum / (float) 3);
+                outbuf1++;
+            }
+            outbuf1 -= 8;
         }
     }
 }
diff --git a/plugins/channel/demoddsd/dsddecoder.cpp b/plugins/channel/demoddsd/dsddecoder.cpp
index 1d1d0e206..fd7ff2429 100644
--- a/plugins/channel/demoddsd/dsddecoder.cpp
+++ b/plugins/channel/demoddsd/dsddecoder.cpp
@@ -18,6 +18,8 @@
 #include <QtGlobal>
 #include "dsddecoder.h"
 #include "dsd_livescanner.h"
+#include "audio/audiofifo.h"
+
 
 DSDDecoder::DSDDecoder()
 {
@@ -25,6 +27,7 @@ DSDDecoder::DSDDecoder()
     initState(&m_dsdParams.state);
 
     m_dsdParams.opts.split = 1;
+    m_dsdParams.opts.upsample = 1; // force upsampling of audio to 48k
     m_dsdParams.opts.playoffset = 0;
     m_dsdParams.opts.delay = 0;
     m_dsdParams.opts.audio_in_type = 0;
@@ -54,6 +57,7 @@ DSDDecoder::DSDDecoder()
 
     m_dsdParams.state.output_buffer = (short *) malloc(1<<18); // Raw output buffer with single S16LE samples @ 8k (max: 128 kS)
     m_dsdParams.state.output_offset = 0;
+    m_dsdParams.state.output_finished = 0;
 
     if (m_dsdParams.state.output_buffer == NULL)
     {
@@ -61,13 +65,17 @@ DSDDecoder::DSDDecoder()
     }
 
     m_dsdParams.state.output_samples = (short *) malloc(1<<19); // Audio output buffer with L+R S16LE samples (max: 128 kS)
-    m_dsdParams.state.output_length = 1<<19; // the buffer size fixed
+    m_dsdParams.state.output_buffers_size = 1<<17; // the buffers size in number of samples: 128 kS
 
     if (m_dsdParams.state.output_samples == NULL)
     {
         qCritical("DSDDecoder::DSDDecoder: Unable to allocate audio L+R buffer.");
     }
 
+    m_dsdParams.state.output_offset = 0;
+    m_zeroBuffer = new short[1<<18]; // 128 kS
+    memset(m_zeroBuffer, 0, sizeof(short) * (1<<18));
+    m_lastNbSamples = 0;
 }
 
 DSDDecoder::~DSDDecoder()
@@ -83,12 +91,45 @@ void DSDDecoder::setInBuffer(const short *inBuffer)
 
 void DSDDecoder::pushSamples(int nbSamples)
 {
-    m_dsdParams.state.input_offset = 0;
-    m_dsdParams.state.input_length = nbSamples;
-
-    if (pthread_cond_signal(&m_dsdParams.state.input_ready)) {
-      printf("DSDDecoder::pushSamples: Unable to signal input ready");
+    if (nbSamples == 0)
+    {
+        m_lastNbSamples = 0;
+        m_dsdParams.state.output_offset = 0; // reset output
     }
+    else
+    {
+        m_dsdParams.state.input_offset = 0;
+        m_dsdParams.state.input_length = nbSamples;
+        m_dsdParams.state.output_finished = 0;
+        m_dsdParams.state.output_length = m_lastNbSamples;
+        m_lastNbSamples = nbSamples;
+
+        if (pthread_cond_signal(&m_dsdParams.state.input_ready)) {
+          printf("DSDDecoder::pushSamples: Unable to signal input ready");
+        }
+    }
+}
+
+void DSDDecoder::popAudioSamples(AudioFifo *audioFifo, bool audioMute)
+{
+    if (audioMute)
+    {
+        uint res = audioFifo->write((const quint8*) m_zeroBuffer, m_dsdParams.state.output_num_samples, 10);
+
+        if (res != m_dsdParams.state.output_num_samples) {
+            qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_dsdParams.state.output_num_samples);
+        }
+    }
+    else if (m_dsdParams.state.output_finished)
+    {
+        uint res = audioFifo->write((const quint8*) m_dsdParams.state.output_samples, m_dsdParams.state.output_num_samples, 10);
+
+        if (res != m_dsdParams.state.output_num_samples) {
+            qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_dsdParams.state.output_num_samples);
+        }
+    }
+
+    m_dsdParams.state.output_finished = 0; // will be done by the next push anyway
 }
 
 void DSDDecoder::start()
@@ -102,6 +143,8 @@ void DSDDecoder::start()
         m_dsdParams.state.dsd_running = 0;
     }
 
+    m_lastNbSamples = 0;
+
     qDebug("DSDDecoder::start: started");
 }
 
diff --git a/plugins/channel/demoddsd/dsddecoder.h b/plugins/channel/demoddsd/dsddecoder.h
index dd55fb826..5e47e938a 100644
--- a/plugins/channel/demoddsd/dsddecoder.h
+++ b/plugins/channel/demoddsd/dsddecoder.h
@@ -20,6 +20,8 @@
 
 #include "dsd.h"
 
+class AudioFifo;
+
 class DSDDecoder
 {
 public:
@@ -27,7 +29,9 @@ public:
     ~DSDDecoder();
 
     void setInBuffer(const short *inBuffer);
+    void pushSample(short sample);
     void pushSamples(int nbSamples); // Push this amount of samples to the DSD decoder thread
+    void popAudioSamples(AudioFifo *audioFifo, bool audioMute);
 
     void start();
     void stop();
@@ -42,6 +46,8 @@ private:
     static void* run_dsd(void *arg);
 
     dsd_params m_dsdParams;
+    short *m_zeroBuffer;
+    int m_lastNbSamples;
 };
 
 #endif /* PLUGINS_CHANNEL_DEMODDSD_DSDDECODER_H_ */
diff --git a/plugins/channel/demoddsd/dsddemod.cpp b/plugins/channel/demoddsd/dsddemod.cpp
index 99c6be73b..0474eb7ed 100644
--- a/plugins/channel/demoddsd/dsddemod.cpp
+++ b/plugins/channel/demoddsd/dsddemod.cpp
@@ -145,52 +145,51 @@ void DSDDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
 				m_scopeSampleBuffer.push_back(s);
 				m_dsdInBuffer[m_dsdInCount++] = sample;
 
-				if (m_running.m_audioMute)
-				{
-	                m_audioBuffer[m_audioBufferFill].l = 0;
-	                m_audioBuffer[m_audioBufferFill].r = 0;
-				}
-				else
-				{
-	                m_audioBuffer[m_audioBufferFill].l = sample;
-	                m_audioBuffer[m_audioBufferFill].r = sample;
-				}
-
-				++m_audioBufferFill;
-
-				if (m_audioBufferFill >= m_audioBuffer.size())
-				{
-					uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
-
-					if (res != m_audioBufferFill)
-					{
-						qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
-					}
-
-					m_audioBufferFill = 0;
-				}
+//				if (m_running.m_audioMute)
+//				{
+//	                m_audioBuffer[m_audioBufferFill].l = 0;
+//	                m_audioBuffer[m_audioBufferFill].r = 0;
+//				}
+//				else
+//				{
+//	                m_audioBuffer[m_audioBufferFill].l = sample;
+//	                m_audioBuffer[m_audioBufferFill].r = sample;
+//				}
+//
+//				++m_audioBufferFill;
+//
+//				if (m_audioBufferFill >= m_audioBuffer.size())
+//				{
+//					uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
+//
+//					if (res != m_audioBufferFill)
+//					{
+//						qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
+//					}
+//
+//					m_audioBufferFill = 0;
+//				}
 
 				m_interpolatorDistanceRemain += m_interpolatorDistance;
 			}
 		}
 	}
 
-	if (m_audioBufferFill > 0)
-	{
-		uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
 
-		if (res != m_audioBufferFill)
-		{
-			qDebug("NFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
-		}
+//	if (m_audioBufferFill > 0)
+//	{
+//		uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
+//
+//		if (res != m_audioBufferFill)
+//		{
+//			qDebug("NFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
+//		}
+//
+//		m_audioBufferFill = 0;
+//	}
 
-		m_audioBufferFill = 0;
-	}
-
-	if (m_dsdInCount > 0)
-	{
-	    m_dsdDecoder.pushSamples(m_dsdInCount);
-	}
+	m_dsdDecoder.popAudioSamples(&m_audioFifo, m_running.m_audioMute);
+    m_dsdDecoder.pushSamples(m_dsdInCount);
 
     if((m_scope != 0) && (m_scopeEnabled))
     {
diff --git a/sdrbase/audio/audiooutput.cpp b/sdrbase/audio/audiooutput.cpp
index 3deb9ba82..c4a0c06ef 100644
--- a/sdrbase/audio/audiooutput.cpp
+++ b/sdrbase/audio/audiooutput.cpp
@@ -179,7 +179,7 @@ qint64 AudioOutput::readData(char* data, qint64 maxLen)
 
 		if (samples != framesPerBuffer)
 		{
-			qDebug("AudioOutput::readData: read %d samples vs %d requested", samples, framesPerBuffer);
+			//qDebug("AudioOutput::readData: read %d samples vs %d requested", samples, framesPerBuffer);
 		}
 
 		for (uint i = 0; i < samples; i++)